Configuring Cisco Voice Over IP, Second Edition, follows some two years after the successful release of its predecessor. On its release, the first edition was at the very leading edge of voice over IP (VoIP) technology and was one of the first texts to be published on this subject. In the short time since the first edition, many aspects of this exciting and expanding technology have changed, and many new protocols and techniques have emerged.This second edition has been fully expanded to include information relating to all these new technologies, with coverage of topics such as Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP). Whether you are a relative newcomer to VoIP networking or currently maintain largescale VoIP networks, this book will prove to be an invaluable addition to your current VoIP information library. Since the time that the first edition of this book was released,VoIP support has increased exponentially and is now more widely deployed in many enterprises around the world.This second edition is intended to serve as a guide to VoIP technology, protocols, and theory.This book is not only a theoretical text, however; it also covers all areas of configuring VoIP with Cisco devices and addresses some of the nontechnical issues relating to VoIP.These include tasks such as preparing business justifications for deploying VoIP networks and preparing a return on investment (ROI) calculation to support your justification.The ability to perform an ROI calculation is a necessary skill in cost justifying a VoIP network. In the early chapters of this book, we look at traditional or legacy voice networks and then analyze in detail the protocols and components that are used in these traditional models. Understanding traditional voice technology is an important and very pertinent aspect to cover, but it is most often overlooked in VoIP texts.Without a solid understanding of the basics of traditional voice networking, it is arguably almost impossible to understand VoIP. Even if VoIP is understood without this baselevel knowledge of traditional voice networking, it is quite likely that certain fundamental aspects of your VoIP deployment will be less than optimal. Many VoIP networks are required to interconnect in some manner with a traditional voice network, such as to the public switched telephone network.An understanding of traditional voicenetworking technology is also invaluable because the historic aspects of voice networking provide an insight into why certain VoIP protocols are designed and operate the way they do.After all, there would be no VoIP without traditional voice After presenting this solid foundation of traditional voicenetworking theory, the book introduces an indepth discussion of VoIP theory. Particular attention is paid to the various protocols that are the cornerstones of any VoIP implementation.The VoIP protocol suites are one of the most complicated aspects of truly understanding VoIP networking. Many network administrators admit that they have only a very basic knowledge of the ways the varying VoIP protocols operate, and often their understanding of such protocols is flawed in some manner.This book intends to dispel some of the myths and mysteries behind VoIP protocols and provide the theory and concepts that underlie these protocols in a clear and concise manner. Naturally, once you develop a sound understanding of VoIP principles, the logical next question is, “What equipment will I require to deploy a VoIP network?”This question is answered in depth in Chapter 4. Cisco has been developing network equipment specifically for VoIP networks for several years, and it will surprise many readers to learn that several of Cisco’s smaller router offerings have been extended to support VoIP networks. Many believe that VoIP requires specialized, expensive hardware and software; this is definitely not the case, and Chapter 4 contains information relating to this misapprehension as well as a detailed overview of current Cisco VoIPcapable devices.We then continue by looking at how these various pieces of hardware can all be configured using the Cisco IOS and the ways that each required configuration is the same, regardless of the hardware platform deployed.Whether you are configuring an analog telephone to connect to a 1700 series router or a 3600 series router, the required configuration commands are the same.This consistent approach to configuring Cisco devices is definitely one of the strengths of using Cisco devices to deploy VoIP. Traditionally, voice network proponents tended to (and sometimes still do) view VoIP technology with some disdain and skepticism; in fact, many believed that VoIP would not be embraced at all.This attitude was mainly due to the fact that early development and deployment of VoIP technology resulted in voice calls that were of very poor quality and highly unreliable at the best of times. Several factors and developments have arisen that have altered this situation.These factors range from the fact www.syngress.comwww.syngress.com that the fundamental VoIP protocols, such as H.323, have improved markedly, to another important factor that has lead to the improved quality of VoIP networks: quality of service (QoS) mechanisms that are now available.These mechanisms are numerous and are all supported on Cisco devices. QoS is an often misunderstood and complicated subject; several books have been written on this subject alone. QoS is so crucial to VoIP networks that many networks that have been deployed without any QoS configuration have subsequently been removed and reverted to traditional voice network technology.This book not only offers a review of the available QoS techniques, it also provides actual configurations of ways to implement several techniques to maintain and improve voice quality. In Chapter 6 we investigate what happens when a voice network fails. Traditionally,VoIP itself was blamed for failed calls in early deployments.VoIP, however, was not generally the cause of poor or failed calls. Underlying network failures tended to be one of the greatest issues regarding VoIP networks.We review some common troubleshooting techniques and then look at a specific technique for troubleshooting network topologies, with particular emphasis on how to troubleshoot VoIP issues. The final two chapters in this book culminate in introducing you to common VoIP case studies.The information in these chapters will provide the answers and configurations to most, if not all, your VoIP requirements.These case studies cover a wide range of VoIP network configuration tasks, ranging from installing a simple analog handset in a router and replacing legacy tieline connections between PBXs to designing and deploying complex dial plans and widescale VoIP solutions.These case studies provide situations in which most of the common VoIP commands are discussed and applied. When you have completed reading this book, you will have a solid foundation of knowledge regarding VoIP networking, with particular emphasis on using Cisco devices.You will understand the protocols that are the essence of VoIP, and you will have as a reference some less commonly documented information regarding financial considerations and how to justify new VoIP deployments.This book provides, in a single reference, all the information you will require to understand, design, deploy, and maintain VoIP networks.This book will become an oftenused tool in your collection of network resources. —Jason Sinclair, CCIE 9100
228_VoIP2e_FM.qxd 8/14/02 4:29 PM Page i solutions@syngress.com With more than 1,500,000 copies of our MCSE, MCSD, CompTIA, and Cisco study guides in print, we continue to look for ways we can better serve the information needs of our readers One way we that is by listening Readers like yourself have been telling us they want an Internet-based service that would extend and enhance the value of our books Based on reader feedback and our own strategic plan, we have created a Web site that we hope will exceed your expectations Solutions@syngress.com is an interactive treasure trove of useful information focusing on our book topics and related technologies The site offers the following features: ■ One-year warranty against content obsolescence due to vendor product upgrades You can access online updates for any affected chapters ■ “Ask the Author” customer query forms that enable you to post questions to our authors and editors ■ Exclusive monthly mailings in which our experts provide 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PM Page iv Syngress Publishing, Inc., the author(s), and any person or firm involved in the writing, editing, or production (collectively “Makers”) of this book (“the Work”) not guarantee or warrant the results to be obtained from the Work There is no guarantee of any kind, expressed or implied, regarding the Work or its contents.The Work is sold AS IS and WITHOUT WARRANTY You may have other legal rights, which vary from state to state In no event will Makers be liable to you for damages, including any loss of profits, lost savings, or other incidental or consequential damages arising out from the Work or its contents Because some states not allow the exclusion or limitation of liability for consequential or incidental damages, the above limitation may not apply to you You should always use reasonable care, including backup and other appropriate precautions, when working with computers, networks, data, and files Syngress Media®, Syngress®,“Career Advancement Through Skill Enhancement®,” “Hack Proofing®,” and “Ask the Author UPDATE®,” are registered trademarks of Syngress Publishing, Inc “Mission Critical™,” and “The Only Way to Stop a Hacker is to Think Like One™” are trademarks of Syngress Publishing, Inc Brands and product names mentioned in this book are trademarks or service marks of their respective companies KEY 001 002 003 004 005 006 007 008 009 010 SERIAL NUMBER 8TPK9H7GYV H8UN7W6CVF 439HF5TS3A Z2B76Z2N9Y UT5R39SC4E X6BUMP7NS6 4EPQ2AKG6R 9BKG8DM5D7 SW4KFAUPFH 5BVM39ZCV6 PUBLISHED BY Syngress Publishing, Inc 800 Hingham Street Rockland, MA 02370 Configuring Cisco Voice Over IP, Second Edition Copyright © 2002 by Syngress Publishing, Inc All rights reserved Printed in the United States of America Except as permitted under the Copyright Act of 1976, no part of this publication may be reproduced or distributed in any form or by any means, or stored in a database or retrieval system, without the prior written permission of the publisher, with the exception that the program listings may be entered, stored, and executed in a computer system, but they may not be reproduced for publication Printed in the United States of America ISBN: 1-931836-64-7 Technical Editor: Jason Sinclair Cover Designer: Michael Kavish Technical Reviewer: Michael E Flannagan Page Layout and Art by: Shannon Tozier Acquisitions Editor: Catherine B Nolan Copy Editor: Darlene Bordwell Developmental Editor: Jonathan Babcock Indexer: Nara Wood Distributed by Publishers Group West in the United States and Jaguar Book Group in Canada 228_VoIP2e_FM.qxd 8/14/02 4:29 PM Page v Acknowledgments We would like to acknowledge the following people for their kindness and support in making this book possible Ralph Troupe, Rhonda St John, Emlyn Rhodes, and the team at Callisma for their invaluable insight into the challenges of designing, deploying and supporting worldclass enterprise networks Karen Cross, Lance Tilford, Meaghan Cunningham, Kim Wylie, Harry Kirchner, Kevin Votel, Kent Anderson, Frida Yara, Jon Mayes, John Mesjak, Peg O’Donnell, Sandra Patterson, Betty Redmond, Roy Remer, Ron Shapiro, Patricia Kelly, Andrea Tetrick, Jennifer Pascal, Doug Reil, David Dahl, Janis Carpenter, and Susan Fryer of Publishers Group West for sharing their incredible marketing experience and expertise Jacquie Shanahan, AnnHelen Lindeholm, David Burton, Febea Marinetti, and Rosie Moss of Elsevier Science for making certain that our vision remains worldwide in scope David Buckland,Wendi Wong, Marie Chieng, Lucy Chong, Leslie Lim, Audrey Gan, and Joseph Chan of Transquest Publishers for the enthusiasm with which they receive our books Kwon Sung June at Acorn Publishing for his support Jackie Gross, Gayle Voycey, Alexia Penny, Anik Robitaille, Craig Siddall, Darlene Morrow, Iolanda Miller, Jane Mackay, and Marie Skelly at Jackie Gross & Associates for all their help and enthusiasm representing our product in Canada Lois Fraser, Connie McMenemy, Shannon Russell, and the rest of the great folks at Jaguar Book Group for their help with distribution of Syngress books in Canada A special welcome to the folks at Woodslane in Australia! Thank you to David Scott and everyone there as we start selling Syngress titles through Woodslane in Australia, New Zealand, Papua New Guinea, Fiji Tonga, Solomon Islands, and the Cook Islands v 228_VoIP2e_FM.qxd 8/14/02 4:29 PM Page vi Contributors Sam Cushway is a Senior Consultant with Callisma where he designs and implements routing and switching solutions for several of Callisma’s clients His specialties include Cisco hardware, strategic network planning, network architecture and design, and network troubleshooting and optimization Sam has 12 years of experience in data communications and his background includes positions as a Senior Network Engineer at TCI Cable and WorldCom Sam is a CCIE candidate with a focus on internetworking and convergence Paul J Fong (CCNP, CCDP) holds a bachelor’s and master’s degree from Stanford University.While pursuing his studies, Paul developed speech recognition software at the Xerox Palo Alto Research Center and published his work in IEEE Transactions on Systems, Man and Cybernetics His background includes positions as an Advisory Systems Analyst at IBM where he developed a network monitoring system for NASA Space Shuttle telemetry, and as a Senior Member of the Technical Staff at MCI where he played a key role in the development of the SRDF-over-IP protocol He is also a contributor to Configuring IPv6 for Cisco IOS (Syngress Publishing, ISBN: 1-928994-84-9) Paul is a member of the Colorado Springs Cisco Users Group and lives in Monument, CO with his wife, Sharon, and their daughter, Shana David Gray (CCNA, CCDA, CCNA-WAN Switching, MCSE, ASE, Brocade Fabric Professional) is a Consultant with Callisma David specializes in IP telephony and convergence, Active Directory design and implementation, storage area networking, service provider networks to include optical design and implementation David’s background includes positions as a Senior Project Engineer for WorldCom, an engineer for Digital Equipment Corporation/Compaq, and as an analyst for a variety of other firms vi 228_VoIP2e_FM.qxd 8/14/02 4:29 PM Page vii Scott M Harris (CCDP, CCNP, MCNE) is a Senior Consultant with Callisma with a focus on internetworking and data/voice/video integration His background includes 14 years of network experience in small, medium, and large network infrastructures He currently provides business consultation on the planning, design, implementation, and Day Two support of Cisco AVVID He has worked with a variety of public and private sector businesses and government institutions on switched LAN and WAN design, implementation, support, and training Larry Keefer, Jr (CCNP, CCDP, MCSE, MCP+I, MCNE, CNE-3, CNE-4, CNE-5, CNE-GW, BCFP, BCSD) is a Senior Consultant with Callisma in Atlanta, GA He currently provides senior-level strategic and technical consulting to Callisma clients throughout the United States His specialties include Cisco routers and LAN switches, Microsoft Windows NT/2000, Novell NDS, IP telephony, storage area network (SAN) planning, design, implementation, and support for large enterprise organizations Larry is a contributing author for Syngress Publishing’s Cisco AVVID & IP Telephony Design and Implementation (ISBN: 1-928994-83-0) Larry’s background includes a position as a Senior Network Engineer/ Team Leader at Rush Creek Solutions (formerly BPI) for enterprise and government organizations in the Rocky Mountain region Larry holds a bachelor’s of Science degree from Illinois Status University and has completed coursework towards a master’s degree in Information Systems from the University of Phoenix Larry currently resides in Woodstock, GA with his family, Carol, Kayla, and Kristin Eric Knipp (CCNP, CCDP, CCNA, CCDA, CIPT, CUSE, MCSE-2000, MCSE, MCP+I) is a Consultant with Callisma Eric specializes in IP telephony and convergence, Cisco routers, LAN switches, Active Directory design and implementation, and network design and implementation He has passed both the CCIE Routing and Switching written exam as well as the CCIE Communications and Services Optical qualification exam Eric is currently preparing to take the CCIE lab later this year Eric’s background includes positions as a Project Manager for a major international law firm and as a Project Manager for Nortel Eric is also a contributing author to Cisco AVVID & IP Telephony Design and Implementation vii 228_VoIP2e_FM.qxd 8/14/02 4:29 PM Page viii (Syngress Publishing, ISBN: 1-928994-83-0), Managing Cisco Network Security, Second Edition (Syngress Publishing, ISBN: 1-931836-56-6), and Configuring IPv6 for Cisco IOS (Syngress Publishing, ISBN: 1-928994-84-9) Estevan Macias (CCIE#5236, CCNA, CCDA, CCNP, CCDP, CIPT, CSS1, CCIE WAN Switching, MCSE, MCP+I) is a Senior Consultant with Callisma He currently provides senior-level strategic and technical consulting to all Callisma clients in the western region of the United States His specialties include Cisco routers and LAN switches, ATM WAN switches, security, design and implementation, strategic network planning, network architecture and design, and network troubleshooting and optimization Estevan’s background includes positions as a Senior Engineer at Timebridge Technologies, and as a Network Engineer at Convergent Communications in the professional services division Estevan holds a master’s of Science in Telecommunications from the University of Denver, and a bachelor’s of Science in Computer Networking from Regis University Charles Riley (CCNP, CSS1, CISSP, CCSA, MCSE, CNE-3) is a Network Engineer with a long tenure in the networking and security fields Charles is the author of the CCNP Routing Study Guide (Osborne McGraw Hill) He has designed and implemented robust networking solutions for several Fortune 500 companies He started with the U.S Army at Fort Huachuca, AZ, eventually finishing his Army stretch as the Network Manager of the Seventh Army Training Command in Grafenwoehr, Germany As a consultant for Sprint in Kansas, Charles developed and managed solutions involving data center design, network migration, storage technologies, and numerous IP routing and transport enterprises for many different customers Charles holds a bachelor’s degree from the University of Central Florida He is fortunate to have the love and encouragement of his wife, René, and his daughter,Tess, who make life so wonderful viii 228_VoIP2e_FM.qxd 8/14/02 4:29 PM Page ix Stuart Ruwet (CCNP–WAN Switching, HPOV Certified Consultant, SNIA Level 1) is a Consultant with Callisma Stuart is a Cisco/StrataCom WAN specialist with broad experience in the telecommunications industry He has worked with a wide variety of networking technologies and platforms including multiple transmission protocols, and network management systems with an emphasis in ATM and IP Prior to working with Callisma, Stuart served as a Senior Network Engineer and Manager of Network Engineering for Convergent Communications, where he was responsible for designing and building a nation-wide ATM network Stuart holds a bachelor’s degree in International Business from the University of Denver He currently lives in Castle Rock, CO with his wife, Meghan, and daughter, Kyla Robert Thorstensen (CCNA, CCNA–WAN, CCDA, CCNP, CCDP, CCIP, CSS1, CIPT, BCFP) is a Consultant with Callisma in Denver, CO and has five years experience in the networking industry Robert has extensive experience with a wide variety of internetworking and optical technologies, protocols, and hardware platforms, with an emphasis on IP telephony, routing, switching, security, and optical networking Robert’s background includes positions as a second tier NOC Engineer at Rhythms NetConnections, and as a Network Engineer at Internet Communications Corporation Robert would like to thank his wife, Lisa, and his three daughters, Brittany, Alyssa, and Aimee, for their patience and understanding on the numerous nights and weekends they would rather have been outside at the park instead of “tip-toeing” around the house Vincent Brian Tillirson is a Consultant with Callisma.Vincent is currently providing technical consulting to a large enterprise banking company in Atlanta, GA His specialties are in WAN/LAN design, implementation and voice deployments.Vincent has participated in the design and implementation of voice and data networks across the United States, Europe and South America.Vincent holds a bachelor’s of Science degree in Computer Science from North Georgia College and University ix 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 531 Index standards, 179 support of ISDN, 257 Cisco Unity System, 41–43 Cisco VIC-2BRI voice interface card, 258 Cisco voice-enabled routers, 67 Cisco VoIP gateways, 141 Cisco VoIP implementation voice modules/cards, 173–178 voice ports, types of, 168–172 VoIP terminology, 202–206 See also specific Cisco hardware names Cisco Web Attendant, 46 Cisco Works 2000, 522 CIST See Cisco IP Telephony (CIST) Class A networks, 115–116, 117 Class B networks, 116, 117 class-based weighted fair queuing (CBWFQ) class maps, defining, 315–316 defined, 157 policies, attaching to interfaces, 318 policies, creation of, 316–318 process of, 291–292 steps of, 314 verifying, 319–320 Class C network, 116–117 Class D address range, 117 class-default, 317–318 Class E address space, 117 class maps, 314, 315–316 class of service, 475 classification, 49 classification and marking category, 520 classification, packet, 320–323 classless interdomain routing (CIDR), 121–122 classless routing protocols, 121, 122 Click-to-Talk See Web Clickto-Talk CO See Central Office (CO) codec bandwidth conservation and, 511 compression technique, 219–220 defined, 203 dial plans and, 235–236 H.323 protocol stack and, 133–134 hybrid, 65 phone calls and, 20–21 trunking and, 252 codec induced delay, 227 codecs, speech, 65–69 See also voice encoding coder See codec command See specific command Common Channel Signaling (CCS), 90–97, 265–266 Compressed Real-Time Transport Protocol (cRTP), 285–287, 301–303, 512 compressed RTP (CR), 285 compressed UDP (CU), 286 compression, 397–399 See also header compression conference calling, 35, 45–46 configuration,VoIP, 212–274 of dial plans/dial peers, 234–248 of gateways/gatekeeper, 268–274 of ISDN for voice, 257–267 of trunking, 249–257 voice port cabling and, 212–219 voice port-tuning commands, 226–234 of voice ports, 219–226 congestion avoidance with WRED, 333–335, 520 congestion management QoS features for, 521 techniques for, 284–285 See also quality of service (QoS) configuration connection command syntax for, 225–226 tie-line variable of, 251 531 for trunking configuration, 249 connection plar-optx command, 252 connection plar string command, 251 connection trunk, 446 connection trunk command, 249 connectionless network protocol, 111 connections, 149–150, 446 continuous bit rate (CBR), 24 control signaling, 132–133 controlled-delay traffic, 295 convergence, 15–16 conversion, analog-to-digital See analog-to-digital conversion conversion, two-to-four wire, 67–68 costs, 13–14 country codes, 223 CPID (calling party identification), 91 CQ See custom queuing (CQ) CR (compressed RTP), 285 cross (C) links, 93 crosstalk, 74 cRTP See Compressed RealTime Transport Protocol (cRTP) CSU (channel service unit), 358–360 CU (compressed UDP), 286 custom queuing (CQ) configuring, 304–308 defined, 157 process of, 288 customer interview, 475 D D (diagonal) links, 93 DARPA (Defense Advanced Research Projects Agency), 114 data link circuit identifier (DLCI), 367–368 data link layer See Layer 2, troubleshooting 228_VoIP2e_indx.qxd 532 8/14/02 4:39 PM Page 532 Index data mining, 47–48 data networks, 14–17 data service unit (DSU), 358 data traffic, 282 data/voice integration See voice/data integration de-jitter, 227–228 debug ATM errors command, 364 debug ATM events command, 364 debug ATM pvcd command, 365 debug commands, 360, 364 debug frame-relay lmi command, 371 debug frame-relay packet command, 371 debug priority command, 380 debug serial interface command, 365 debug serial packet command, 365 decimal value, binary numbering, 114–115 decoder See codec default class, 317–318 Defense Advanced Research Projects Agency (DARPA), 114 delay call quality and, 522 defined, 203 link fragmentation/ interleaving for, 299–300 satellite effect, 513 voice port-tuning commands and, 226–228 delay-sensitive traffic, 295 demarcation point, 61 deny route map, 322 design plan extending, 495–496, 504–505 reviewing, 480–481 for single-router VoIP network, 479–480 design principles, tie-line replacement, 429–432 design review, 480–481 destination pattern dial-peers, troubleshooting, 395 dial plan and, 235, 503–504 in router configuration, 484, 491 symbols used in, 238 trunking and, 252, 253 detail recording, packetized voice, 47–48 device discovery and registration, 135 diagonal (D) links, 93 dial-digit interpretation, 252–253 dial peer in design plan, 479–480 IP precedence configuration with, 321 IP precedence for, 293 See also POTS dial peer dial-peer commands for fax over VoIP, 244 for POTS, 240–241 for VoFR, 244–245 for VoIP, 242–243 Dial Peer Configuration mode, 491, 497, 512 dial peer statements, 483–484, 497–499 dial peer tag, 503–504 dial-peer voice type command, 437–438 dial peers bandwidth conservation and, 512 defined, 203 for interoffice voice communication, 507–508, 509–510 troubleshooting, 395–399 dial plan defined, 203 extending, 494–495, 503–504 in general, 384–385 intraoffice, creating, 474–478 mapping, 480 planning of, 385–387 selecting for packetized voice, 39–40 show dialplan number xxxx command, 387–388 spreadsheet, 432, 434–435 dial plans/dial peers, configuring call legs and, 236–237 configuring dial peers, 239–246 creating, implementing dial plan, 237–239 designing/planning dial plan, 235–236 dial plan, defined, 234 direct inward dialing, 246–248 number expansion and, 246 dial-pulse signaling, 73, 81–82 dial tone, 79, 479 dialogue,TCAP session, 96 DID/FXS voice interface card, 178 DID ports, 224–225, 233–234 differential pulse code modulation (DPCM), 66 differentiated service, 156, 283 digit translation, 205 digital loop carrier (DLC), 100–101 Digital Number Identification Service (DNIS), 246 digital signal processors (DSP), 173 digital transmission in general, 85 ISDN signaling, 87–89 time-division multiplexing, 85–87 digital trunks, 487–488 digital voice interfaces, 260–261 dimensioning, 84 direct inward dialing (DID), 246–248, 476 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 533 Index direct voice trunking, 252–253 discovery and registration call stage, 134–135 DLC (digital loop carrier), 100–101 DLCI (data link circuit identifier), 367–368 DNIS (Digital Number Identification Service), 246 DNS server, 270 documentation, 419, 423 DPCM (differential pulse code modulation), 66 drop policy, 317 drop wire, 61 ds0-group command, 437 DS1 interface, 95 DSP (digital signal processors), 173 DSU (data service unit), 358 dual-tone multifrequency (DTMF) signaling, 73, 82–83 E E&M acronym, 352 E&M interface, 170–172 E&M ports configuring, 222–224 fine-tuning, 231–234 in PBX-VoIP migration, 488–492 troubleshooting, 352–353, 392–395 E&M signaling gateways/gatekeeper, 405–407 overview of, 77–80 PBX and, 77, 394 trunk seizure, 402–405 wiring schemes, 401–402 E1 voice connectivity, 172 ear and mouth See E&M signaling earth and magnet See E&M signaling echo defined, 204 described, 226, 228 testing, 513 EIGRP See Enhanced Interior Gateway Routing Protocol (EIGRP) email, 41–42 encapsulation frame-relay command, 370 encoding, analog-to-digital, 67 end stations, 124 endpoint-to-endpoint signaling, 143–144 endpoints of H.323 protocol, 127, 128 media stream and, 141–142 in MGCP connection, 149–150 Enhanced Interior Gateway Routing Protocol (EIGRP), 507, 508, 510 equipment testing/verifying, 358–360 troubleshooting, 346–348 Erlang B, 84–85 Ethernet 1/0 interface, 268–269 Ethernet cables, 357 Ethernet port configuration of, 496–497, 498 for single-router VoIP network, 478, 479 event record, 513 extended (E) links, 94 extensibility, SIP, 145 extensions, 476–477 F F (fully associated) links, 94 fair-queue command defined, 291 for LFI configuration, 336 for WFQ configuration, 312 fast switching, 294 fax merged networks and, 41–42 over VoIP, 244 packet telephony and, 10 533 FDM (frequency-division multiplexing), 86 feeder cable, 61 FHR (fixed hierarchical routing), 64 fiber optic cabling, 71–72 first in, first out (FIFO), 285 fixed delay, 227 fixed hierarchical routing (FHR), 64 fixed keyword, 228 fixed-length subnet masking (FLSM), 121 flag, 292–293 flow, 298 flow-based WFQ, 290 flow classification, CBWFQ, 291 FLSM (fixed-length subnet masking), 121 Foreign Exchange Office Interface (FXO), 169–170, 253 foreign exchange station (FXS) port for interoffice voice communication, 508–509 in PBX-VoIP migration, 488–489 for single-router VoIP network, 479, 481 Foreign Exchange Station Interface (FXS), 168–169 See also FXS/FXO ports Foresight feature, 298 fragmentation delay, 336 frame relay defined, 204 for toll-bypass, 17–21 troubleshooting, 367–371 Frame Relay encapsulation configuration example of, 303 enabling cRTP with, 301 Frame Relay links, 522 frame-relay traffic-rate command, 332 Frame Relay traffic shaping (FRTS) 228_VoIP2e_indx.qxd 534 8/14/02 4:39 PM Page 534 Index configuring, 331–332 described, 298 verifying, 333 frequency, 74 frequency-division multiplexing (FDM), 86 frequency modulation, 98 frequency shifting key (FSK), 98 FRTS See Frame Relay traffic shaping (FRTS) fully associated (F) links, 94 FXO (Foreign Exchange Office Interface), 169–170, 253 FXS/FXO ports configuration of, 212–213, 220–222 fine-tuning, 229–231 signaling errors, troubleshooting, 399–401 voice, 351–352, 388–392 G G.729, 24 G.7XX codec, 134 gap analysis, 179 gatekeeper, 127, 204 See also specific gatekeeper model gatekeeper ID, 269 gatekeeper-routed call signaling, 130, 137–138 gateway command, 268 gateways defined, 204 E&M signaling, 405–407 SIP and, 146 See also specific gateway models generic traffic shaping (GTS), 298, 331 ground start lines, 80–81 ground-start signaling, 253, 399, 400–401 H H.224 protocol, 132 H.225 message, 138 H.245 control signaling, 132–133 Open Logical Channel Acknowledge message, 140 protocol, 124, 139–140 session, 143 H.248, 150–151 H.26X codec, 134 H.323, 204 H.323 call setup logical channel setup, 139–140 media stream/media control flows, 141–142 phases of, 137–138 H.323 call stages call setup, 137–142 call termination, 142–143 discovery and registration, 134–135 endpoint-to-endpoint signaling, 143–144 interzone call placement, 136–137 intrazone call placement, 135–136 H.323 gatekeeper defined, 127 in H.323 call stages, 135, 136–137 services of, 129–130 H.323 gateway configuring, 268–269 described, 127 services of, 128–129 zone bandwidth, 297, 325–329 H.323 ID addresses, 270 H.323 protocol, 126–131 components, 127 function of, 126–127 gatekeeper, 129–130 gateways, 128 importance of, 110 Multipoint Control Units, 130–131 terminals, 128 when to use, 151–152 H.323 protocol stack, 131–134 H.323 standard, 16 H.323v.1, 126 hardware for single-router VoIP network, 478–479 VoIP migration and, 418, 422 header compression for bandwidth conservation, 512 command for, 302 configuration example, 302–303 with cRTP, 285–286 high-bandwidth traffic, 290 high complexity, 219–220 high-usage routes, 64 holding area, 287 hoot-n-holler service, 446 hopoff command, 274 host address, 116–117, 119–121 host portion, IP address, 118–119 hybrid codecs, 65 hyperterminal session, 482 I immediate-start signaling, 255–257 impedance, 74 implementation services carrier and intercarrier VoIP, 11–13 Cisco IP Telephony, 8–10 corporate multimedia, 10–11 toll bypass, 6–8 in-band signaling, 91 information collection compilation of, 420–421 for design purposes, 417–420 in general, 416–417 presentation of, 423–428 for tie-line replacement, 421–423 integrated service, 155–156, 283 Integrated Services Digital Network See ISDN integration of SIP, 145 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 535 Index voice/data, 451 interactive voice response (IVR), 40–41 intercarrier VoIP, 11–13 interface applying priority list to, 310 attaching policy to, 318–319 LFI verification and, 336–337 multilink, 335–336 point-to-multipoint, 369–370 states, 367 Interface Configuration mode, 512–513 International Telecommunications Union (ITU), 87 Internet, 112, 126 Internet Protocol Device Control (IPDC), 148 Internet Protocol (IP) H.323 protocol stack and, 131 in OSI reference model, 111, 112 Internet Service Provider (ISP), 100, 115 interoffice voice communication, configuring VoIP for design, extending, 504–505 dial plan, extending, 503–504 routers, configuring, 505–511 interoperability, SIP, 145 interzone call placement, 136–137, 328 interzone parameter, 325 intraoffice dial plan, creating, 474–478 intraoffice/interoffice VoIP scenarios, 474–522 interoffice voice communication, configuring VoIP for, 503–511 intraoffice dial plan, creating, 474–478 multirouter intraoffice VoIP network, configuring, 494–502 PBX and VoIP networks, merging, 486–494 QoS, applying to VoIP network, 520–522 single-router VoIP network, configuring, 480–486 single-router VoIP network, designing, 478–480 VoIP network, tuning, 511–519 intrazone call placement, 135–136 invite message, 147 IP See Internet Protocol (IP) IP address classes, 115–117 in general, 114–115 mapping, 368–369, 375 network portion, 118–119 private address space, 117–118 subnetting, 118–121 supernetting, 121–122 IP network protocols, 112–114 IP packet, 378 IP phone, 479 ip policy route-map command, 322 IP precedence configuration of, 321 congestion avoidance with WRED, 333–334, 335 described, 293–294 levels, 315–316 IP routing, 376–377 ip rsvp bandwidth command, 323 IP RTP Priority configuring, 329–331 described, 292 queuing, 157 ip rtp priority command, 297 IP telephony CIPT and, maintenance, 35 PBX replacement with, 34–38 IP transport, 11–13 IPDC (Internet Protocol Device Control), 148 535 IPv4 standard, 114 ISDN, 257–267 ISDN BRI voice interface card, 178 ISDN BRI voice ports, 258–260 ISDN configuration for voice of CAS signaling, 264–265 of CCS signaling, 265–266 Cisco platform support, 257 of ISDN BRI voice ports, 258–260 of ISDN PRI voice ports, 260–263 of Q.931 support, 263 of QSIG, 264 of T-CCS, 266–267 ISDN PRI circuit configuring CCS with, 265–266 configuring QSIG with, 264 configuring with Q.931, 263 ISDN PRI voice ports, 260–263 ISDN signaling, 87–89 ISDN User Part (ISUP), 95–96 ISO reference model, H.323 standards, 125 ISP (Internet Service Provider), 100, 115 itp rtp priority command, 329–330 IVR (interactive voice response), 40–41 J jitter, 204, 227 K keepalive intervals, 367 key stakeholders, 418, 422 key system, 62–63 L latency, 154 Layer 1, troubleshooting cabling, 348–351 CO problems, 355–356 defined, 346 228_VoIP2e_indx.qxd 536 8/14/02 4:39 PM Page 536 Index equipment, powerup to operating state, 346–348 PBX problems, 353–354 ports, 351–353 testing/verifying cable, 357–358 testing/verifying Cisco router interfaces, 360–365 testing/verifying communications equipment, 358–360 Layer 2, troubleshooting ATM, 372–375 defined, 346 frame relay, 367–371 in general, 365 serial interfaces, 365–367 Layer 3, troubleshooting bandwidth reservation protocols, 381–384 defined, 346 in general, 375 IP addressing/routing, 376–377 IP QoS, 377–379 queuing, 379–380 LCSE (Logical Channel Signaling Entity), 139 LDAP (Lightweight Directory Access Protocol), 41 LED status, 217–219 LFI See link fragmentation/ interleaving (LFI) Lightweight Directory Access Protocol (LDAP), 41 line finder, 80 link efficiency mechanisms, 520 link fragmentation/interleaving (LFI), 299–300, 335–337 links, Frame Relay, 522 LLC (Logical Link Control) sublayer, 365 LLQ (low latency queuing), 379–380, 521 location confirm (LCF), 137 location request (LRQ), 137 logging, 48 logical channel setup, 133, 139–140 Logical Channel Signaling Entity (LCSE), 139 Logical Link Control (LLC) sublayer, 365 long distance, 35, 69 loop start lines, 80–81 loop start signaling, 399–400 loopback diagnostic command, 373 low-bandwidth traffic, 290 low latency queuing (LLQ), 379–380, 521 LRQ (location request), 137 M maintenance, IP telephony, 35 make interval, 82 mapping, dial plan, 480 mark probability denominator (MPD), 334 Match statement, 294, 322 MC (multipoint controller), 131 MC3810 router, 215–216 MCM (Cisco Multimedia Conference Manager), 38–39 MCS (Cisco Media Convergence Server), MCUs See multipoint control units (MCUs) mean opinion score (MOS), 67, 430–431 Media Access Control (MAC) sublayer, 365 Media Gateway Control Protocol (MGCP) connections, 149–150 in general, 148–149 MeGaCo/H.248, 150–151 Session Initiation Protocol and, 145 status of, 111 when to use, 152 media gateway controllers (MGCs), 148–151 Media Gateway Protocol (MGCP), 204–205 media gateways (MGs), 148–149 media stream, media control flows, 141–142 medium complexity, 219, 220 MeGaCo/H.248, 150–152 Message Transfer Part (MTP), 94–95 messages, SIP, 147–148 methodology testing/verification, 463–467 troubleshooting, 345–346 MGCP See Media Gateway Control Protocol (MGCP) MGCs (media gateway controllers), 148–151 MGs (media gateways), 148–149 Microsoft NetMeeting software, 128 MMUSIC (Multiparty Multimedia Session Control) working group, 144 mode of propagation, 71–72 modem, 98–100 modular command-line (MQC), 513 modularity, SIP, 145 modulation, 97 See also amplitude modulation MOS See mean opinion score (MOS) moves, 35 MP (multipoint processor), 131 MPD (mark probability denominator), 334 MQC (modular commandline), 513 MTP (Message Transfer Part), 94–95 multilink interface, 335–336 multilink PPP, 335–337 multimedia, corporate, 10–11 Multiparty Multimedia Session Control (MMUSIC) working group, 144 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 537 Index multipoint calls, 150 multipoint control units (MCUs), 127, 130–131 multipoint controller (MC), 131 multipoint processor (MP), 131 multirouter intraoffice VoIP network, configuring design, extending, 495–496 dial plan, extending, 494–495 routers, configuring, 496–499 testing/verification, 499–502 music on hold, 46 N NANP (North American Numbering Plan), 475 NAT (Network Address Translation), 38 network access servers, 149 administrator, 293 analysis, 417 Class A, 115–116, 117 Class B, 116, 117 Class C, 116–117 components, analog, 61–65 convergence, 282 management, 419–420 merging, 41–43 planning, 64–65 switching, 227 troubleshooting, 356–357 Network Address Translation (NAT), 38 NM-1V/2V voice network module, 173–174 NM-HDV voice network module, 174–176 no digit-strip command, 438 “noise”, 76 North American Numbering Plan (NANP), 475 northamerica keyword, 221, 223 num-exp command, 246 number expansion command for, 246 configuring for four-digit dialing, 510–511 dial plan and, 235 number extension, 205 O octects, 116–117, 118–119 off-net/on-net calls, 477 one-to-one connection, 150 Open Systems Interconnection (OSI) model, 344, 345 options message, 148 OSI reference model, 111, 112 out-of-band signaling, 91 overhead, cRTP, 285, 287 P PABX See private automatic branch exchange (PABX) packet classification, 292–293, 320–323 packet size, 112 packet-switched voice networks, 152–158 packet voice/data module (PVDM), 182 packetization delay, 227 packetized voice call routing and, 38–39 detail recording/data mining with, 47–48 dial plan selection for, 39–40 implementation services, 6–13 interactive voice response and, 40–41 quality of service of, 48–49 as replacement for PBX, 37–38 TAPI integration and, 43–44 toll-bypass designs, 13–37 transcoders and, 48 transfer/forward/conference capabilities of, 45–46 unified messaging with Cisco Unity System, 41–43 Web Click-to-Talk and, 43–44 See also Voice over Internet Protocol (VoIP) packets CBWFQ and, 291–292 537 in custom queuing, 288 link fragmentation/ interleaving, 299–300 policy-based routing for, 294 priority queuing and, 289 queuing, 287 WRED for, 298–299 padding, 24 passive keyword, 301 payback period, 33 PBR See policy-based routing (PBR) PBX See Private Branch Exchange (PBX) PCM See pulse code modulation (PCM) peer type, 235 performance, 153–154 permanent virtual circuit (PVC), 372–373, 374 phase modulation, 98 phase shift, 75 phone bills, 30–31 phone system abuse, 80 ping test, 377 plain old telephone service (POTS), 205 planning See information collection plar option, 225 plar-optx option, 226 PLAR-OPX command, 252 playout-delay command, 228 point-to-multipoint interface, 369–370 point-to-point leased lines, 27–29 point-to-point links, policer, 521 policies, CBWFQ, 316–318 policy-based routing (PBR), 294, 322–323 policy maps, 314 port, 351–353, 479 DID, 224–225, 232–234 port number, 329–330 228_VoIP2e_indx.qxd 538 8/14/02 4:39 PM Page 538 Index port-number-range guideline, 330 port numbering on 1700 series, 213–214 on 2600 and 3600 series, 214–215 on 7200 series, 216 on AS5x00 series, 216–217 on MC3810 series, 215–216 POTS dial peer configuring, 239–241, 498–499, 509 troubleshooting, 395 versus voice network dial peer, 236–237 POTS (plain old telephone service), 205 power denial, 253 ppp multilink interleave command, 336 PQ See priority queuing (PQ) precedence, 378–379 predictive service traffic, 295 prefixes See technology prefix Primary Rate Interface, 88 priority-group command, 310 priority list, 308–311 priority-list command, 308–309 priority queuing (PQ) configuring, 308–312 defined, 156–157 process of, 288–289 with WFQ, configuring, 329–331 priority, RTP, 297–298 private address space, 117–118 private automatic branch exchange (PABX), 416 See also Voice over Internet Protocol (VoIP), migrating to Private Branch Exchange (PBX), 354–355 described, 62 E&M signaling and, 77 ground start lines and, 80 interconnect trunks, 486–488 problems, 353–354 replacement of with IP Telephony, 34–38 system, 123–124, 258–260 tie-line replacement and, 13 VoIP implementation and, 4–6 and VoIP networks, merging, 486–494 private-line automatic ringdown (PLAR) configuration of, 251–252 option for, 221 trunking and, 446–447 protocol stack, SS7, 94–97 protocol translation, H.323 gateway, 128 provisioning, 49 proxy server, 146 public switched telephone network (PSTN) as backup, 16 defined, 205 dial plans and, 384 DID port configuration and, 224 H.323 standard and, 16 North American, dial plan of, 234 port connection with, 213 VoIP implementation and, pulse code modulation (PCM), 65–66, 67, 511 PVC (permanent virtual circuit), 372–373, 374 PVDM (packet voice/data module), 182 Q Q.931 request, 123–124 Q.931 support, 263 QSIG, 264 Quality of Service (QoS) applications for, 153–154 defined, 153 importance in VoIP networks, 157–158 importance of, 111 IP, 377–379 IP measures of, 23 levels of, 154–156 queuing versus, 156–157 role in packet-switched voice networks, 152–158 services availability and, 25–26 on VoFR, 18 VoIP and, 48–49 VoIP networks and, 520–522 Quality of Service (QoS) configuration bandwidth, maximizing, 284–285, 300 call admission control, 295–297, 325–329 class-based weighted fair queuing, 314–320 congestion avoidance with WRED, 335–337 of cRTP, 285–287, 301–303 custom queuing, 304–308 IP precedence, 293–294 link fragmentation/ interleaving, 299–300 overview of, 282–283 packet classification, 292–293, 320–323 policy-based routing, 294 priority queuing, 308–312 priority queuing with WFQ, 329–331 queuing, 287–292, 303–304 RSVP, 294–295, 323–325 RTP priority, 297–298 traffic shaping, 298, 331–333 weighted fair queuing, 312–314 weighted random early discard, 298–299 queue-limit command, 317 queue-limit parameter, 309–310 queue-list command, 304 queue sizes, 305–306 queuing configuring, 303–320 methods of, 156–157 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 539 Index plan for, 284–285 QoS tool, 49 troubleshooting, 379–380 types of, 287–292 variable delay in, 227 R random-detect command, 317, 333 random-detect precedence command, 334 RAS (Registration, Admission, and Status), 129, 133 rate-sensitive traffic, 295 Real-Time Control Protocol (RTCP), 124, 141 Real-Time Transport Protocol (RTP) bandwidth reservation and, 381–383 function of, 133 header compression, 285–286 priority, 297–298 RED algorithm, 298 redirect server, 146 reference points, ISDN, 88–89 refraction, 71 register message, 148 registered jack (RJ) cables, 349–351 registrar server, 146 registration, 135 Registration, Admission, and Status (RAS), 129, 133 residential gateways, 149 Resource Reservation Protocol (RSVP) bandwidth reservation and, 383–384 configuring, 323–325 configuring for QoS, 521 function, types of, 294–295 IP precedence and, 293, 294 Quality of Service and, 155–156 resources Cisco Connection Online (CCO), 478 for dial plan implementation, 495 for QoS, 111 return on investment (ROI) building of and payback period, 33 case study: PBX/IP Telephony, 34–37 solution design, 32 telephony, cost review, 29–31 of toll-bypass, 29–37 RFC 1918, 117–118 RFC 2508, 285 RJ-11 cabling, 212 RJ-1CX cabling, 212–213 RJ-48C connector, 170 RJ cables See registered jack (RJ) cables ROI See return on investment rollback procedures, 464 route-map statements, 294, 322 router configurations examples, 439–445, 447–451, 453–463 routers configuring for PBX-VoIP migration, 488–492 configuring for QoS, 521 for DID system, 246–247 for interoffice voice communication, 505–511 in multirouter intraoffice VoIP network, 496–502 port numbering on Cisco routers, 213–219 scalability, 28 signaling from PBX to, 123–124 for single-router VoIP network, 478–486 See also specific router model routing See call routing routing protocols, 125 RSVP See Resource Reservation Protocol (RSVP) RTCP See Real-Time Control Protocol (RTCP) 539 RTP See Real-Time Transport Protocol (RTP) S satellite effect, 513 scalability of cRTP, 303 of dial plan, 474, 475 of Session Initiation Protocol, 145 of VoIP, 27–28 SCCP (Signaling Connection and Control Part), 96–97 SCP (service control point), 91–92 security, 419, 423 serial interfaces, 365–367 serial line, 301 serialization, 227 serialization delay, 205, 299–300 service control point (SCP), 91–92 service models, 154–156 service policy attaching to interface, 318–319 creating, 316–318 verifying, 319–320 service switching point (SSP), 91–92 Session Initiation Protocol (SIP) benefits of, 145 components, 146–147 function of, 144–145 messages, 147–148 proxy servers and, 146 status of, 110–111 when to use, 152 session parameter, 325 session target, 235, 395 Set statement, 294 set statements, policy routing, 322 SGCP See Skinny Gateway Control Protocol (SGCP); Simple Gateway Control Protocol (SGCP) shaper, 521 228_VoIP2e_indx.qxd 540 8/14/02 4:39 PM Page 540 Index show atm pvc module/port command, 374 show command, 256–257, 322–323, 326–328, 360 show controllers command, 358, 362–363, 373 show diagnostic command, 361–362 show dial-peer command, 486 show frame-relay map command, 371 show frame-relay pvc command, 370–371 show interface atm (module/port) command, 373–374 show interface command performance testing and, 320 for priority queuing, 311 for troubleshooting, 363–364 for troubleshooting ATM, 373 for verifying custom queuing, 306–307 for WFQ configuration, 313–314 show interface serial x command, 365–367 show ip interface command, 377 show ip protocols command, 377 show ip rsvp installed command, 323–324 show ip rsvp interface command, 324 show policy-map command, 319 show policy-map interface command, 319–320 show ppp multilink command, 336 show queuing command, 307–308 show queuing fair command, 314 show queuing priority command, 311–312 show queuing random-detect command, 334–335 show running command, 394–395 show running-config command, 319 show version command, 347–348, 360–361, 505 show voice port 1/1/0 command, 247–248 show voice port command, 492–494 sidetone, 60 signal phase, 75 signal transfer point (STP), 91–92 signaling CAS, 264–265 control, 132–133 DTMF, 73, 82–83 endpoint-to-endpoint, 143–144 in-band, 91 ISDN, 87–89 out-of-band, 91 QoS features for, 521 between router and PBX, 123–124 VoIP, 124–125 wink-start versus immediatestart, 233–234, 255–257 See also analog signaling; call control signaling; E&M signaling Signaling Connection and Control Part (SCCP), 96–97 signaling errors, troubleshooting, 399–407 signaling links, SS7, 92–94 signaling points, SS7, 91–92 signaling rate, 79 Signaling System (SS7) in general, 90–91 protocol stack, 94–97 signaling links, 92–94 signaling points, 91–92 Simple Gateway Control Protocol (SGCP), 148 single-router VoIP network configuring, 480–486 designing, 478–480 SIP See Session Initiation Protocol (SIP) Skinny Gateway Control Protocol (SGCP), 205 slots, router, 213–217 software, 419, 423 source encoding, 65, 66–69 speech codecs, 65–69 See also voice encoding SS7 See Signaling System (SS7) SSP (service switching point), 91–92 stakeholders See key stakeholders standards, cabling, 72–73 starting-rtp-port guideline, 330 statistics, cRTP, 302 STP (signal transfer point), 91–92 streamed data transfer, 113 string, 225 subinterfaces, 369 subnet mask, 118–119, 121–122 supernetting, 121–122 supervisory disconnect, 253–255 supervisory disconnect tone, 253 support, IP telephony, 35 switches, 10, 255 switchhook, 59 T T-CSS (Transparent Common Channel Signaling), 266–267 T1, 445 T1/E1 modules, 260–263 T1 voice connectivity, 172 T.38 standard, 244 TA568A/B wiring schemes, 350 tag switching, 125 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 541 Index tail drop use of, 291 WRED and, 299, 335 tandeming, 172 TAPI See Telephony Application Programming Interface (TAPI) TCAP (Transaction Capabilities Application Part), 96 TCP See Transmission Control Protocol (TCP) TCP/IP, 114, 118 TDM (time-domain multiplexing), 205–206 technology prefix, 269, 271–274 telephone system operation, 59–61 telephone system switches, 61–62 Telephone User Part (TUP), 97 telephones, analog, 61 telephony cost review, 29–31 solutions design, 32 See also voice telephony principles Telephony Application Programming Interface (TAPI), 43–44 terminal equipment type, 88 terminals, H.323, 127, 128 testing cable, 357–358 calls, 513–519 Cisco router interfaces, 360–365 equipment, communications, 358–360 multirouter intraoffice VoIP network, 499–502 QoS in VoIP network, 521–522 router-to-PBX telephone connectivity, 492–494 single-router VoIP network, 485–486 voice systems, 521–522 VoIP network tuning, 513–519 tie-line option, 221, 225 tie-line port, 488 tie-line replacement, 464–466 tie-line replacement, configuring, 435, 436–445 tie-line replacement, designing, 429–435 tie-lines in router configuration, 491 toll-bypass and, 13–14 tie lines variable, 251 time-division multiplexing, 85–87 time-domain multiplexing (TDM), 205–206 toll bypass, 6–8 toll-bypass designs return on investment of, 29–37 tie-line replacement and, 13–14 using ATM for, 21–27 using Frame Relay for, 17–21 using point-to-point leased lines for, 27–29 voice/data networks, merging of, 14–17 toll trunk line, 68 ToS (Type of Service) field, 378–379 total parameter, 325 tracing, 48 traffic, 295 traffic classes, CBWFQ, 291–292 traffic conditioning, 521 traffic-shape rate command, 331 traffic shaping configuring, 331–333 features of, 298 Transaction Capabilities Application Part (TCAP), 96 transcoders, 48 541 Transmission Control Protocol (TCP) function of, 112–113 H.323 protocol stack and, 132 WRED works with, 299 Transparent Common Channel Signaling (T-CSS), 266–267 troubleshooting basic methodology, 345–346 dial peers, 395–399 dial plans, 384–388 in general, 344–345 Layer 1, 346–364 Layer 2, 365–375 Layer 3, 375–384 signaling errors, 399–407 voice ports, 388–395 VoIP commands, 396 VoIP issues, 397 trunk connection, 446 trunk, digital, 487–488 trunk option, 225 trunk seizure delay-start signaling, 404 immediate-start signaling, 403 signaling, troubleshooting, 404–405 wink-start signaling, 402–403 trunk signaling, 123–124 trunking analog, 77, 487–488 direct voice, 252–253 trunking, advanced options, case studies, 447–451 options overview, 446–447 scenarios, configuring, 446 testing, 466–467 trunking, configuring direct voice versus dial-digit interpretation, 252–253 PLAR, 251–252 PLAR-OPX, 252 of supervisory disconnect, 253–255 tie lines, 251 228_VoIP2e_indx.qxd 542 8/14/02 4:39 PM Page 542 Index trunks, 249–250 wink-start signaling versus immediate-start signaling, 255–257 trunking gateway, 149 trunks configuring, 249–250 PBX interconnect and, 486–488 TUP (Telephone User Part), 97 two-to-four wire conversion, 67–68 Type of Service (ToS) field, 378–379 U unified messaging, 41–43 universal port card, AS5350, 217 unshielded twisted-pair (UTP) cable, 70 user agent client (UAC), 146 user agent server (UAS), 146 user agent (UA), 146 User Datagram Protocol (UDP) function of, 113–114 H.323 protocol stack and, 132 user-to-network interface, H.323 terminals, 128 V V.92 in modem connections, 99 throughput, factors that affect, 100–101 VAD See voice activity detection (VAD) variable delays, 227 variable-length subnet masking (VLSM), 121 verifying and testing, voice/data integration, 463–467 VIC See voice interface card (VIC) VIC-2E/M voice interface card, 176 VIC-2FXO voice interface card, 177 VIC-2FXS voice interface card, 177 video conferencing with Cisco IP Telephony, 11 multipoint control units for, 130–131 video, H.323 protocol, 126 virtual connection, 113, 249 VLSM (variable-length subnet masking), 121 VNM See voice network module (VNM) VoATM See voice over Asynchronous Transfer Mode (VoATM) vocoders, 65, 67 VoFR See voice over Frame Relay (VoFR) voice activity detection (VAD), 206, 512 voice card See voice interface card (VIC) voice conferencing, 130–131 voice/data integration case studies of, 451–463 rationale for, 451 verifying/testing, 463–467 voice encoding, 65–69 voice interface card (VIC), 173, 176–178 voice interface card (VIC) modules, 212–213, 218 voice interface, digital, 200–201 voice module See voice network module (VNM) voice network, 14–17 voice network dial peer, 236–237 voice network module (VNM) Cisco routers support for, 214 Cisco VoIP implementation and, 173–176 LED location on, 217 voice port cabling and, 212 voice over Asynchronous Transfer Mode (VoATM) advantages of, 26 defined, 206 dial peers and, 245–246 in general, IP QoS and, 23 toll bypass networks and, 22 voice over ATM gateways, 149 voice over Frame Relay (VoFR) defined, 206 dial-peer setup, 243–245 toll bypass networks and, 17–18 VoIP implementation and, 6–7 when to use, 19–21 where to use, 18 voice over HDLC (VoHDLC), 27 voice over Internet Protocol (VoIP) defined, 206 design guidelines, 429 dial-peer, 241–243 interoffice voice communication, 503–511 issues for troubleshooting, 397 packets, 17 terminology, 202–206 troubleshooting commands, 396 See also packetized voice voice over Internet Protocol (VoIP), migrating to information collection for, 416–428 integrated voice/data and, 451–463 tie-line replacement, configuring, 435–445 tie-line replacement, designing, 429–435 trunking scenarios, configuring advanced, 446–451 verifying and testing, 463–467 voice over Internet Protocol (VoIP) network applying QoS to, 520–522 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 543 Index merging with PBX network, 486–494 multirouter intraoffice, configuring, 494–502 single-router, 478–486 tuning, 511–519 voice port dial plan and, 235 in modified dial plan, 503–504 parameter commands in, 489–490 router configuration and, 483 troubleshooting, 351–353 troubleshooting E&M, 392–395 troubleshooting FXS/FXO, 388–392 voice port cabling, 212–219 on 1700 series, 213–214 on 2600 and 3600 series, 214–215 on 7200 series, 216 on AS5x00 series, 216–217 in general, 212–213 LEDs and, 217–219 on MC3810 series, 215–216 voice port-tuning commands delay and echo concepts, 226–228 for E&M ports, 231–234 for FXS/FXO ports, 229–231 voice ports, configuring connection command for, 225–226 DID ports, 224–225 E&M ports, 222–224 FXO or FXS ports, 220–222 steps for, 219–220 voice ports, types E&M interface, 170–172 E1/T1 voice connectivity, 172 FXO, 169–170 FXS, 168–169 voice technologies, 3–6 voice Telephony principles analog network components, 61–65 analog signaling, 73–85 analog systems, 58–61 analog -to-digital conversion, 97–101 analog voice encoding, 65–69 cabling for analog, 69–73 Channel Associated Signaling (CAS), 89–90 Common Channel Signaling (CCS), 90–97 digital transmission, 85 ISDN signaling, 87–89 time-division multiplexing, 85–87 voice traffic delay and echo in, 226–228 priority queuing with WFQ and, 292 QoS for, 282 voicemail, 41–42 VoIP signaling and voice transport protocols in general, 110–111 H.323 call stages, 134–144 H.323 protocol, 126–131 H.323 protocol stack, 131–134 IP networks overview, 111–122 Media Gateway Control Protocol, 148–152 QoS role in packet-switched voice networks, 152–158 Session Initiation Protocol, 144–148 signaling, addressing, routing, 122–125 VWIC-2MFT-T1 multiflex trunk interface card, 177–178 543 W WAN link bandwidth conservation and, 511, 512, 513 program for, 522 WAN (Wide Area Network), 9–10 Web Click-to-Talk, 43–44 weighted fair queuing (WFQ) configuring, 312–314 defined, 156 priority queuing with, 292 priority queuing with WFQ, 329–331 process of, 289–290 weighted random early discard (WRED) CBWFQ and, 291 configuring, 333–335 described, 298–299 WFQ See weighted fair queuing (WFQ) wholesale dial, 11 Wide Area Network (WAN), 9–10 wink-start signaling for DID ports, 233–234 versus immediate-start signaling, 255–257 wiring schemes,TA568A/B, 350 WRED See weighted random early discard (WRED) Z zone local command, 270–271 zone management, H.323 gatekeeper, 130 zone prefixes, 270, 272 zone remote command, 270, 271 228_VoIP2e_indx.qxd 8/14/02 4:39 PM Page 544 SYNGRESS SOLUTIONS… AVAILABLE NOW! ORDER at www.syngress.com Cisco AVVID and IP Telephony Design & Implementation Many lessons have been learned from the rapid adoption of Cisco’s AVVID IP Telephony product line Companies must cost effectively integrate “old world” technologies with the new, engineer packet transport networks for quality of service, simultaneously bolster traditional telephony and IP skills, and continuously exercise proper risk management This book will help readers overcome the inherent complexities of IP telephony so that its promise can be fully realized ISBN: 1–928994–83–0 Price: $69.95 USA, $108.95 CAN AVAILABLE NOW! 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ORDER at www.syngress.com Managing Cisco Network Security, Second Edition Information security has become an extremely important topic over the past few years In today’s environment the number of touch points between an organization’s information assets and the outside world has drastically increased Millions of customers interact via Web sites, employees and partners connect via Virtual Private Networks, applications are outsourced to Application Service Providers (ASPs) and wireless LANs are regularly deployed Cisco Systems has placed a high priority on security and offers a wide range of security products Managing Cisco Network Security, Second Edition is important to anyone involved with Cisco networks, as it provides practical information on using a broad spectrum of Cisco’s security products ISBN: 1–931836–56–6 Price: $59.95 USA, $92.95 CAN solutions@syngress.com Document3 4/3/02 4:04 PM Page ... Signaling Configuring ISDN for Voice Configuring ISDN BRI Voice Ports Configuring ISDN PRI Voice Ports Configuring Q.931 Support Configuring QSIG Configuring CAS Configuring CCS Configuring T-CCS Configuring. .. Introduction to Voice Over IP and Business Justifications Introduction Introduction to Voice Over IP General Overview of Voice Technologies Today’s VoIP Possibilities The PBX Reality The VoIP Bandwagon... President and CEO, Callisma www.syngress.com 228_VoIP2e_intro.qxd 8/14/02 4:40 PM Page xxix Introduction Configuring Cisco Voice Over IP, Second Edition, follows some two years after the successful